Mirrors/overte-JulianGro
3
0
Fork 0
mirror of https://github.com/JulianGro/overte.git synced 2025-04-25 21:15:07 +02:00
Code Issues Projects Releases Packages Wiki Activity Actions

Allow local echo when audio input and output devices have different sample rates

Browse source
This commit is contained in:
Ken Cooke 2019-05-30 08:59:45 -07:00
parent b0ace1c098
commit cabe9eab81
2 changed files with 30 additions and 14 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

43
libraries/audio-client/src/AudioClient.cpp
View file

@ -291,6 +291,7 @@ AudioClient::AudioClient() :
_inputToNetworkResampler(NULL), _inputToNetworkResampler(NULL),
_networkToOutputResampler(NULL), _networkToOutputResampler(NULL),
_localToOutputResampler(NULL), _localToOutputResampler(NULL),
_loopbackResampler(NULL),
_audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT), _audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT),
_outgoingAvatarAudioSequenceNumber(0), _outgoingAvatarAudioSequenceNumber(0),
_audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this), _audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this),
@ -762,6 +763,11 @@ void AudioClient::stop() {
qCDebug(audioclient) << "AudioClient::stop(), requesting switchOutputToAudioDevice() to shut down"; qCDebug(audioclient) << "AudioClient::stop(), requesting switchOutputToAudioDevice() to shut down";
switchOutputToAudioDevice(QAudioDeviceInfo(), true); switchOutputToAudioDevice(QAudioDeviceInfo(), true);
if (_loopbackResampler) {
delete _loopbackResampler;
_loopbackResampler = NULL;
}
// Stop triggering the checks // Stop triggering the checks
QObject::disconnect(_checkPeakValuesTimer, &QTimer::timeout, nullptr, nullptr); QObject::disconnect(_checkPeakValuesTimer, &QTimer::timeout, nullptr, nullptr);
QObject::disconnect(_checkDevicesTimer, &QTimer::timeout, nullptr, nullptr); QObject::disconnect(_checkDevicesTimer, &QTimer::timeout, nullptr, nullptr);
@ -1085,13 +1091,6 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
return; return;
} }
// NOTE: we assume the inputFormat and the outputFormat are the same, since on any modern
// multimedia OS they should be. If there is a device that this is not true for, we can
// add back support to do resampling.
if (_inputFormat.sampleRate() != _outputFormat.sampleRate()) {
return;
}
// if this person wants local loopback add that to the locally injected audio // if this person wants local loopback add that to the locally injected audio
// if there is reverb apply it to local audio and substract the origin samples // if there is reverb apply it to local audio and substract the origin samples
@ -1108,21 +1107,30 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
} }
} }
// if required, create loopback resampler
if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
qCDebug(audioclient) << "Resampling" << _inputFormat.sampleRate() << "to" << _outputFormat.sampleRate() << "for audio loopback.";
int channelCount = (_inputFormat.channelCount() == 2 && _outputFormat.channelCount() == 2) ? 2 : 1;
_loopbackResampler = new AudioSRC(_inputFormat.sampleRate(), _outputFormat.sampleRate(), channelCount);
}
static QByteArray loopBackByteArray; static QByteArray loopBackByteArray;
int numInputSamples = inputByteArray.size() / AudioConstants::SAMPLE_SIZE; int numInputSamples = inputByteArray.size() / AudioConstants::SAMPLE_SIZE;
int numLoopbackSamples = (numInputSamples * OUTPUT_CHANNEL_COUNT) / _inputFormat.channelCount(); int numInputFrames = numInputSamples / _inputFormat.channelCount();
int numLoopbackFrames = (numInputFrames * _outputFormat.sampleRate() + _inputFormat.sampleRate() - 1) / _inputFormat.sampleRate();
int numLoopbackSamples = numLoopbackFrames * OUTPUT_CHANNEL_COUNT;
loopBackByteArray.resize(numLoopbackSamples * AudioConstants::SAMPLE_SIZE); loopBackByteArray.resize(numLoopbackSamples * AudioConstants::SAMPLE_SIZE);
int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data()); int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data()); int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
// upmix mono to stereo possibleResampling(_loopbackResampler,
if (!sampleChannelConversion(inputSamples, loopbackSamples, numInputSamples, _inputFormat.channelCount(), OUTPUT_CHANNEL_COUNT)) { inputSamples, loopbackSamples,
// no conversion, just copy the samples numInputSamples, numLoopbackSamples,
memcpy(loopbackSamples, inputSamples, numInputSamples * AudioConstants::SAMPLE_SIZE); _inputFormat.channelCount(), OUTPUT_CHANNEL_COUNT);
}
// apply stereo reverb at the source, to the loopback audio // apply stereo reverb at the source, to the loopback audio
if (!_shouldEchoLocally && hasReverb) { if (!_shouldEchoLocally && hasReverb) {
@ -1892,15 +1900,22 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
_outputDeviceInfo = QAudioDeviceInfo(); _outputDeviceInfo = QAudioDeviceInfo();
} }
// cleanup any resamplers
if (_networkToOutputResampler) { if (_networkToOutputResampler) {
// if we were using an input to network resampler, delete it here
delete _networkToOutputResampler; delete _networkToOutputResampler;
_networkToOutputResampler = NULL; _networkToOutputResampler = NULL;
}
if (_localToOutputResampler) {
delete _localToOutputResampler; delete _localToOutputResampler;
_localToOutputResampler = NULL; _localToOutputResampler = NULL;
} }
if (_loopbackResampler) {
delete _loopbackResampler;
_loopbackResampler = NULL;
}
if (isShutdownRequest) { if (isShutdownRequest) {
qCDebug(audioclient) << "The audio output device has shut down."; qCDebug(audioclient) << "The audio output device has shut down.";
return true; return true;

1
libraries/audio-client/src/AudioClient.h
View file

@ -390,6 +390,7 @@ private:
AudioSRC* _inputToNetworkResampler; AudioSRC* _inputToNetworkResampler;
AudioSRC* _networkToOutputResampler; AudioSRC* _networkToOutputResampler;
AudioSRC* _localToOutputResampler; AudioSRC* _localToOutputResampler;
AudioSRC* _loopbackResampler;
// for network audio (used by network audio thread) // for network audio (used by network audio thread)
int16_t _networkScratchBuffer[AudioConstants::NETWORK_FRAME_SAMPLES_AMBISONIC]; int16_t _networkScratchBuffer[AudioConstants::NETWORK_FRAME_SAMPLES_AMBISONIC];